[..] |
ASTERISK-16000: app_queue crashes asterisk |
ASTERISK-16001: [patch] Manager cookies isn't compatible with rfc2109 |
ASTERISK-16002: Remote dialplan execution stops randomly over IAX2/SIP. |
ASTERISK-16003: [patch] asterisk dsp always reports detected DTMF length to be 0ms |
ASTERISK-16004: Call transfer from Voicemail or Queue Application result Asterisk to crash. (SIP REFER) |
ASTERISK-16005: [patch] Memory leak in manager.c |
ASTERISK-16006: [regression] X-Lite disconnects because of RTCP timeout when Local channel and MeetMe/F involved |
ASTERISK-16007: module unload res_smdi.so errors |
ASTERISK-16008: [patch] Improve usability of logging information for misconfigured contexts |
ASTERISK-16009: [patch] Provisioning failure |
ASTERISK-16010: startup fails with libspeex.so.1.2.0 |
ASTERISK-16011: Blind transfer problem |
ASTERISK-16012: [regression] attended transfered calls appearance in the queue_log |
ASTERISK-16013: MixMonitor fails to record atxfer calls |
ASTERISK-16014: [patch] segfault caused by error in SQL |
ASTERISK-16015: It crashes severl times a day under heavy load |
ASTERISK-16016: Asterisk crash on module reload func_odbc |
ASTERISK-16017: [patch] it crashes handling dtmf relay |
ASTERISK-16018: It crashes in rtp_timeput disconnect function |
ASTERISK-16019: [patch] Pretty Config Files |
ASTERISK-16020: [patch] Proposed patch to associate ActionId with UniqueId earlier when originating a call |
ASTERISK-16021: Inband DTMF Double digits being sent. |
ASTERISK-16022: [patch] Multiple SIP NOTIFY packets sent with LDAP realtime peers |
ASTERISK-16023: [patch] UDP ports not freed/ports leaking |
ASTERISK-16024: Call gets disconnected after approx 20 seconds of ringing |
ASTERISK-16025: [patch] Long hints make asterisk crash |
ASTERISK-16026: sip_pvt does not have association with sip_peer for INVITEs |
ASTERISK-16027: [patch] Update Introduction to README.txt with some more verbosity and links |
ASTERISK-16028: ARA - error if fields are empty (not NULL) |
ASTERISK-16029: 1.4.31rc1 Deadlock: Tried and failed to get Lock #1 (chan_dahdi.c): MUTEX 1063 pri_grab &pri->lock 0xe09684 |
ASTERISK-16030: [patch] "queue reset stats" erroneously clears wrapuptime configuration |
ASTERISK-16031: [patch] Freenum-in-a-can dialplan correction for failing pattern match |
ASTERISK-16032: [patch] Crash when using Background() in Macro called by M() option to Dial() |
ASTERISK-16033: NTT Japan will ignore SIP packets with ;received= set in Via SIP header |
ASTERISK-16034: [patch] Failed to register peers from realtime config |
ASTERISK-16035: [patch] audiohook: translation to slinear 8khz regardless of samplerate |
ASTERISK-16036: [patch] [Regression] non-root make install PREFIX=/tmp fails |
ASTERISK-16037: DAHDI FXS channels: CALLERID(num) and CALLERID(name) are empty after first hangup |
ASTERISK-16038: getting warning message every 4 seconds |
ASTERISK-16039: [patch] [regression] Segmentation fault in check_rtp_timeout |
ASTERISK-16040: [patch] Segmentation fault in scheduled event |
ASTERISK-16041: atxfer *2 channel dahdi FXS no hangup |
ASTERISK-16042: BAD ROUND TIME FOR ANSWEREDTIME |
ASTERISK-16043: [patch] bypass "contactdeny" with nat=yes |
ASTERISK-16044: [patch] The heap data structure can't cope with a removal and reinsert |
ASTERISK-16045: tcptls.c:254 ast_tcptls_server_root: Accept failed: Bad file descriptor |
ASTERISK-16046: MOH not playing when phone transfer softkey pressed |
ASTERISK-16047: [patch] Allow disconnect feature before a call is bridged in branch 1.6.2 |
ASTERISK-16048: Asterisk segfault xml_translate manager api |
ASTERISK-16049: UPGRADE-1.6.txt doesn't mention insecure=very/yes removal |
ASTERISK-16050: SIP attended transfer broken |
ASTERISK-16051: SIP Anonymous Call: h323 to SIP |
ASTERISK-16052: [patch] [regression] No T.140 text for outbound SIP calls |
ASTERISK-16053: [patch] update chan_sip.c for AST_CLI_YESNO and AST_CLI_ONOFF |
ASTERISK-16054: AMI Originate does not return a "Reason" when called with Async = 0 |
ASTERISK-16055: Asterisk Crashes with segfault |
ASTERISK-16056: [patch] chan_iax2 ignores the port in an SRV record |
ASTERISK-16057: [patch] [backport for 1.4] Autopause only pauses member on queue where timeout took place |
ASTERISK-16058: SIP device shows "IN USE" / Busy when it is not |
ASTERISK-16059: [patch] endless cycle with ast_channel->lock held leeds to complete system stuck |
ASTERISK-16060: app_jack connects two ports, but complains it doesn't |
ASTERISK-16061: Register statement in sip.conf does not accept slash in user part (valid per RFC 3261) |
ASTERISK-16062: chan_sip.c: Peer 'XXX' is now UNREACHABLE |
ASTERISK-16063: Registration process retries even though OK has been returned from the provider |
ASTERISK-16064: [patch] [regression] asterisk 1.4.31 crashes (segmentation fault) on sip to sip call via iax2 trunk |
ASTERISK-16065: [patch] Eliminate compiler warning in app_voicemail.c |
ASTERISK-16066: [patch] pthread_rwlock_timedwrlock autoconf test is wrong |
ASTERISK-16067: [patch] module reload cdr_odbc.so produces an error about not being able to register ODBC method with CDR. |
ASTERISK-16068: language resets during blind transfer |
ASTERISK-16069: [patch] Sequencing numbers of IMAP email messages is incorrect |
ASTERISK-16070: asttest fails to compile in centos 5.3 |
ASTERISK-16071: AST_EXT_LIB_CHECK doesn't use [extra cflags] |
ASTERISK-16072: channelstats packet loss is reported wrong |
ASTERISK-16073: [patch] Busy/hangup tone detection on Voicemail recording |
ASTERISK-16074: SIP NOTIFY bombardment after hangup voicemail app |
ASTERISK-16075: [patch] ./configure --with-pri=LIBPATH does not use libpri from LIBPRI_PATH |
ASTERISK-16076: pri_cc_enable not in current libpri |
ASTERISK-16077: [patch] MeetMe 'L' and 'S' ignore 'C' option |
ASTERISK-16078: [patch] FXS Polarity Reversal on Hangup or Answer |
ASTERISK-16079: asterisk 1.6.2.7 does not properly renumber/delete voicemail files when the maxmsg is exceeded |
ASTERISK-16080: sip.conf: parkinglot=abc << this variable exists still after removing |
ASTERISK-16081: [regression] asterisk sip becomes unresponsive, astobj2.c:115 INTERNAL_OBJ: bad magic number 0x0 for 0x9f0a828 |
ASTERISK-16082: Asterisk randomly stops accepting calls in g729 format |
ASTERISK-16083: Playback(file,noanswer) in Asterisk-1.6.2.7 (1.6.2.8rc1 too) does not work properly |
ASTERISK-16084: Asterisk crash after Hangup app |
ASTERISK-16085: Asterisk 1.6.2.6 randomly crashes |
ASTERISK-16086: [patch] Computation of Content-Length is wrong under some circumstances |
ASTERISK-16087: Random crash in chan_unistim |
ASTERISK-16088: [patch] Additional logging for testsuite when skipping a test |
ASTERISK-16089: [patch] logger.conf file with invalid syntax |
ASTERISK-16090: Asterisk crash on ODBC /usr/lib/libodbc.so.1 and /lib/i686/cmov/libc.so.6 |
ASTERISK-16091: [patch] segfault on logging |
ASTERISK-16092: [regression] Operator context execution broken |
ASTERISK-16093: Asterisk doesn't compile anymore on Mac OS 10.6.3 |
ASTERISK-16094: [patch] [regression] CDRs being written where they were not before after revision 258670 |
ASTERISK-16095: ExternalIVR 'S' command sometimes results in an Asterisk segmentation fault |
ASTERISK-16096: [patch] Fix some doxygen warnings |
ASTERISK-16097: [patch] [regression] flooding /var/spool/asterisk/outgoing/xxxxx.call: No such file or directory |
ASTERISK-16098: No ringtone when going from queue to dial-command |
ASTERISK-16099: [patch] SIP Directed pickup can result in dead channel. |
ASTERISK-16100: 1.6.1.18 : app_dial times out when call forward set on the phone, forward channel still observes original dial timeout |
ASTERISK-16101: possible hint state deadlock |
ASTERISK-16102: ENUMLOOKUP not returning queried for entry |
ASTERISK-16103: meetme music on hold stop working and RTP stop flowing |
ASTERISK-16104: MeetMe destroy the conference if X option is included and MeetMeAdmin can no longer find the last conference room |
ASTERISK-16105: ast_close_fds_above_n off-by-one's |
ASTERISK-16106: [regression] PlayTones() does not produce any tones in 1.6.2 (works in 1.6.1) |
ASTERISK-16107: Asterisk generates responses with missing Via headers |
ASTERISK-16108: DTMF (info) is no longer passed through after 180 seconds |
ASTERISK-16109: applicationmap does not support applications that use more than 1 argument (1.6.x does not accept pipes) |
ASTERISK-16110: problem with receivefax |
ASTERISK-16111: [patch] Asterisk deadlock |
ASTERISK-16112: [patch] Autopause on busy issue |
ASTERISK-16113: [patch] Bad error message if Gosub or GosubIf to bad address |
ASTERISK-16114: Bad magic number |
ASTERISK-16115: [patch] problem with ringinuse=no, queue members receive sometimes two calls |
ASTERISK-16116: [patch] 'stutter' and 'dialrecall' indications for NZ missing comma's between tones. |
ASTERISK-16117: Asterisk transcodes audio "internally" when calling between two SIP peers when it should not |
ASTERISK-16118: [patch] Redirecting ;1 side of local channel during optimisation causes double free of ;1 side and crash |
ASTERISK-16119: atxfer, one way sound, codecs |
ASTERISK-16120: Race condition causes manager session event list to underflow causing null pointer de-ref and crash. |
ASTERISK-16121: [patch] strange extension pattern matching |
ASTERISK-16122: jabberrecieve application is missing |
ASTERISK-16123: [patch] realtime shows 2x fullcontact if delimited by ';' |
ASTERISK-16124: [patch] ast_readstring (multiple DTMF input) doesn't transmit silence to the caller even if transmit_silence=yes |
ASTERISK-16125: [patch] [regression] DAHDI analog FXS port segfaults after dialling 2nd DTMF digit |
ASTERISK-16126: [patch] [regression] Progress in band error (don't send RTP packets) |
ASTERISK-16127: no cdr records if no cdr.conf since 1.6.2 branch |
ASTERISK-16128: Poor man's find-me/follow-me dialplan implementation |
ASTERISK-16129: iax2-parser eating memory if not LOW_MEMORY set. |
ASTERISK-16130: [patch] res_ldap.conf points md5secret to RealmedPassword, but the schema uses AstAccountRealmedPassword |
ASTERISK-16131: I can not make Hangup from manager & PHP... |
ASTERISK-16132: StopMixMonitor crashes at the end of the recording, when initiated by the callee. |
ASTERISK-16133: [patch] Notify event keep-alive is responded with bad event |
ASTERISK-16134: [patch] long waiting (over 2 min) for cdr_tds running "SELECT 1 FROM cdr" when asterisk starting with aournd 3mil cdr in table |
ASTERISK-16135: CPU usage very high when putting call on MOH |
ASTERISK-16136: AGISTATUS bug in Asterisk 1.6.1.8 |
ASTERISK-16137: Proposed method of avoiding registration probing bots |
ASTERISK-16138: [patch] Memory corruption from iksemel |
ASTERISK-16139: Calltoken's and IAX2 realtime configuration |
ASTERISK-16140: Modules will not load |
ASTERISK-16141: [patch] Race conditition in app_meetme leads to crash |
ASTERISK-16142: [patch] latest trunk doesn't compile |
ASTERISK-16143: [patch] pgsql realtime module not using local unix socket |
ASTERISK-16144: [patch] AGISTATUS bug in Asterisk 1.6.2.7 |
ASTERISK-16145: [patch] [regression] r263292 breaks zaptel compatibility so Asterisk fails to make. |
ASTERISK-16146: [patch] typo in res_agi.c documentation |
ASTERISK-16147: When caller exits due to EXITEMPTY in app_queue EXITUNAVAIL is written to queue_log. |
ASTERISK-16148: ERROR[11190]: chan_dahdi.c:11859 process_dahdi: Unknown signalling method 'pri_net' |
ASTERISK-16149: MWI NOTIFY is not being sent to phones subscribed to events |
ASTERISK-16150: Writeformat Slin instead Alaw after attendend Transfer when using Answer before Dial without Moh |
ASTERISK-16151: [patch] [regression] Incoming overlap dialing no longer works after rev 203304 |
ASTERISK-16152: Asterisk cannot see Jack compiled from source -- DEBIAN |
ASTERISK-16153: [patch] RTP directmedia is broken in some cases |
ASTERISK-16154: [patch] [regression] audio delay when bridging calls related to timestamp mismatch |
ASTERISK-16155: Speakerphone LED Update Fail after call termination. |
ASTERISK-16156: [patch] DEADLOCK_AVOIDANCE can actually generate dealocks |
ASTERISK-16157: [patch] MoH not restarted after end of conference announcement is played |
ASTERISK-16158: [patch] member goes in auto-paused status if calling call get queue timeout |
ASTERISK-16159: Caller ID in atxfer |
ASTERISK-16160: [patch] Voicemail envelope doesn't correct read date/time when voicemail stored in IMAP |
ASTERISK-16161: [patch] Deadlock in chan_local.c causes crash. |
ASTERISK-16162: [patch] ss_thread calls pri_grab without lock during overlap dial |
ASTERISK-16163: IAX2 bug causes asterisk hangup, flooding log with '__ast_queue_frame: Exceptionally long voice queue length queuing to' message |
ASTERISK-16164: MGCP Zombie Channel |
ASTERISK-16165: segfault error. Asterisk crash. |
ASTERISK-16166: It crashes randomly |
ASTERISK-16167: [patch] queue reload members does not work |
ASTERISK-16168: No transfer or MOH activated for AMI D4 circuit |
ASTERISK-16169: app_queue's compare_weight can be called in contexts that don't lock the global queues lock |
ASTERISK-16170: Asterisk crashes on sip_realtime using MySQL via ODBC |
ASTERISK-16171: Outgoing call file sometimes crashes Asterisk |
ASTERISK-16172: Problems with siren14 codec; problems with siren7 sound files. |
ASTERISK-16173: Bad defaulting in chan_skype.c |
ASTERISK-16174: Segfault after launching JACK_HOOK from AMI |
ASTERISK-16175: [patch] Australian Accent core sounds submission |
ASTERISK-16176: function SHARED broken |
ASTERISK-16177: segfault in for loop embedded switch statement |
ASTERISK-16178: [patch] dialplan add failing to create context if context doesn't exist |
ASTERISK-16179: Memory Leak : main/manager.c |
ASTERISK-16180: [patch] DAHDI bridge can't fax pass through |
ASTERISK-16181: [patch] transfer from queue problem (loopbug) |
ASTERISK-16182: [patch] [branch] unprivileged (non-root) installation of Asterisk |
ASTERISK-16183: Strange warning from dahdi channels |
ASTERISK-16184: [patch] crash when From header URI misses "sip:" |
ASTERISK-16185: Asterisk cores @ 110 calls |
ASTERISK-16186: Context names ending on space are not processed correctly. |
ASTERISK-16187: Replace soxmix with sox -m |
ASTERISK-16188: dialplan remove a@b tab completion seg fault |
ASTERISK-16189: [patch] priexclusive in chan_dahdi.conf ignored when reloading dahdi module |
ASTERISK-16190: "make -j X" fails |
ASTERISK-16191: Stuck channel after asterisk feature attended transfer |
ASTERISK-16192: [patch] app_swift text-to-speech engine |
ASTERISK-16193: I looked at it. What am I supposed to be seeing? |
ASTERISK-16194: [patch] In agent calls recording, CDR userfield is not updated when the call is transferred |
ASTERISK-16195: Asterisk will not Start - Just crashes. |
ASTERISK-16196: App Dial "crashes" when received a 484 (Address Incomplete) from a SIP Trunk |
ASTERISK-16197: [patch] "setvar" can add multiple variables with the same name to a channel |
ASTERISK-16198: [patch] SIP display-name needed to be empty for Avaya IP500 |
ASTERISK-16199: requirecalltoken does not work with Realtime |
ASTERISK-16200: [patch] Patch adds REAL duration and billsec to CDR |
ASTERISK-16201: [patch] valid IP address in RTP offer when Asterisk is attached to several networks |
ASTERISK-16202: [patch] obvious deadlock |
ASTERISK-16203: Asterisk crashes upon loading the res_jabber module. |
ASTERISK-16204: Please look at the trace. |
ASTERISK-16205: Peer does not hang up when caller hangup while app_dial is executing - Deadagi |
ASTERISK-16206: When cross-compiling, menuselect is built for target, not host architecture |
ASTERISK-16207: Errors on Asterisk console |
ASTERISK-16208: Dahdi callerid name is no passed |
ASTERISK-16209: Asterisk can't match peer |
ASTERISK-16210: [patch] rt(c)p set debug ip takes wrong argument |
ASTERISK-16211: [patch] sort=alpha does not start at position 0 |
ASTERISK-16212: [patch] "joinempty = ringing" doesn't work |
ASTERISK-16213: [patch] parser mangles #include |
ASTERISK-16214: [patch] mistakes in the 1.4.19 core-sounds-en.txt file |
ASTERISK-16215: [patch] Crash in dsp.c when entering digits from SpeechBackground |
ASTERISK-16216: [patch] Replace old stub functions with new optional_api functions |
ASTERISK-16217: [patch] [regression] Constantly heard "beep" on PC speaker and Asterisk use maybe 10 time more CPU when CONSOLE=yes |
ASTERISK-16218: [patch] cdr_pgsql does not detect when a table is not found |
ASTERISK-16219: [patch] 3 way transfer - avoid deadlock |
ASTERISK-16220: [patch] bashism in configure script |
ASTERISK-16221: [patch] editing files in main/editline does not ensure rebuild of libedit.a |
ASTERISK-16222: getting warning message every 4 seconds |
ASTERISK-16223: [patch] [regression] el_gets re-entered after el_end (at shutdown) causes segfault |
ASTERISK-16224: Asterisk crash if caller is busy |
ASTERISK-16225: Impossible to record a call when the originator makes a transfer |
ASTERISK-16226: [patch] response_refer() does not have a default case, so a 400 final response stalls in the Transfer() application |
ASTERISK-16227: [patch] app_recordkey - new record application that returns key pressed |
ASTERISK-16228: Certain C-style comments cause large blocks of code to be ignored |
ASTERISK-16229: Seg faulting when stopping asterisk |
ASTERISK-16230: calls show some seconds on my cdrs - calls show minutes on my carrier's cdrs!! |
ASTERISK-16231: logger.conf does not create subdirectories or give warnings about them missing |
ASTERISK-16232: Dial with MOH |
ASTERISK-16233: [patch] "dahdi show channels group" auto-completion bug |
ASTERISK-16234: [patch] [regression] Segmentation fault in scheduled event |
ASTERISK-16235: [patch] Bug in reporting queue hold time |
ASTERISK-16236: [patch] wrapuptime not respected properly by app_queue: a queue member with setup wrapuptime gets a call when inappropriate |
ASTERISK-16237: [patch] Speex Wideband Support |
ASTERISK-16238: [patch] [regression] MusicOnHold don't play if MeetMe Room is stored in mysql db (realtime) |
ASTERISK-16239: [patch] know the context of the parking call |
ASTERISK-16240: [patch] ShowDialPlan does not end with ShowDialPlanComplete Event if manager events are off |
ASTERISK-16241: problem with SIP during reboot |
ASTERISK-16242: MSG_OOB flag on HANGUP packet misused |
ASTERISK-16243: Channel h323 does not load |
ASTERISK-16244: G729 codecs not being released |
ASTERISK-16245: Bruteforce hack |
ASTERISK-16246: chan_h323 crashes Asterisk |
ASTERISK-16247: [patch] typos fixed in channelvariables.tex |
ASTERISK-16248: Include empty folder in exensions.conf - weird results |
ASTERISK-16249: [patch] prematuremedia and progress inband |
ASTERISK-16250: [patch] udptl parameters differ with and without udptl.conf |
ASTERISK-16251: [regression] Operator context execution broken |
ASTERISK-16252: app_meetme documentation for option w[(secs)] is called out as W[(secs)] (incorrectly capitalized) |
ASTERISK-16253: chan_mobile (revision 421) crash Asterisk |
ASTERISK-16254: Norwegian language - wrong word |
ASTERISK-16255: [regression] Dialing to another Asterisk system fails (codec negotiation error) |
ASTERISK-16256: [patch] Documentation updates for 'Developer branches and branch merging' |
ASTERISK-16257: [patch] EventList start and EventList end manager variables do not match |
ASTERISK-16258: [patch] Brief lagginess on IAX2 channels is fatal |
ASTERISK-16259: app_confbridge never plays enter/leave sounds. |
ASTERISK-16260: [patch] app_confbridge ignores option 's' to present menu. |
ASTERISK-16261: [patch] handle DAHDI_EVENT_REMOVED on a D-Channel |
ASTERISK-16262: Asterisk seg faults after incoming OOH323 call fails due to congesstion |
ASTERISK-16263: [patch] New Zealand indications.conf tones incorrect |
ASTERISK-16264: Locked PRI spans |
ASTERISK-16265: [patch] Severe clicking, popping and talkoff when inbound call from SIP |
ASTERISK-16266: [patch] Adding useragent in realtime database |
ASTERISK-16267: Crash after queue/agents module reload |
ASTERISK-16268: [patch] Linksys SPA94x keep-alive reply replies to wrong address |
ASTERISK-16269: Attended transfers not working with IP phones |
ASTERISK-16270: [patch] Add Speex Wideband sample frames |
ASTERISK-16271: [patch] queue reload clears queue statistics |
ASTERISK-16272: [patch] unload mod is not complete |
ASTERISK-16273: Asterisk crashes when processing invalid tif file |
ASTERISK-16274: [patch] Set dynamic parameters for queue and queue members. |
ASTERISK-16275: asterisk not hanging up |
ASTERISK-16276: All Calls Originated with .call files using Local Channels are logged with NO ANSWER Disposition. |
ASTERISK-16277: eventwhencalled=yes does not generate any events |
ASTERISK-16278: g729 codec leak / licenses remain used when no calls active |
ASTERISK-16279: [patch] TCP connection will not be closed, if device do a reregister |
ASTERISK-16280: [regression] DTMF transfers stop working for queue agents idle > 5 minutes |
ASTERISK-16281: "Received a DCN from remote after sending a page." while using SendFAX as an aswering machine |
ASTERISK-16282: REGISTER attempts with stale nonce |
ASTERISK-16283: [patch] Localizattion for spanish in say.conf |
ASTERISK-16284: [patch] comebacktoorigin=no returns to flattened channel, not extensions s, for parked call timeout |
ASTERISK-16285: [patch] Wrong check for privilege in Originate action |
ASTERISK-16286: [patch] Realtime erase username when Unavailable |
ASTERISK-16287: Cancelling a connection does not clear retransmissions |
ASTERISK-16288: Memory leak on reload |
ASTERISK-16289: Asterisk crashes when 2 simultaneous chan_mobile FXOs are used |
ASTERISK-16290: Upgrade Asterisk from 1.4.x to latest 1.6 - - - starting asterisk 1.6 gives continuosly crash + restart |
ASTERISK-16291: chan_oss.c with video support want load whan ffmpeg missing |
ASTERISK-16292: [patch] voicemail_odbc_postgresql step 11 typo |
ASTERISK-16293: Asterisk crashes on startup |
ASTERISK-16294: X-LITE disconnects due to RTCP timeout (1.4.33.1) |
ASTERISK-16295: Parsing error in extconfig.conf results in seg fault |
ASTERISK-16296: MOH stops working after a few seconds and P2P bridging restarts despite still on hold! |
ASTERISK-16297: [patch] Reference in extconfig should force driver preload |
ASTERISK-16298: [patch] SRTP (SRTP unprotect: authentication failure) |
ASTERISK-16299: [patch] Asterisk "locks up" the system when an external process is called from the 'h' extension with a lower priority than Aste |
ASTERISK-16300: [patch] IPv6 support - SIP and RTP |
ASTERISK-16301: [patch] Tweak to voicemail to put rdnis in msgxxxx.txt file |
ASTERISK-16302: Asterisk crashes after redirect |
ASTERISK-16303: [patch] cdr->src variable is not set anymore in destination channels |
ASTERISK-16304: [patch] [regression] RFC 2833 frame out of order detection does not properly handle numeric overflow |
ASTERISK-16305: app_dial.c: dial_exec_full Unable to create channel type 'SIP' (cause 20 - unknown) |
ASTERISK-16306: T.38 UDPTL Port Negotiation Fails |
ASTERISK-16307: Unable to transfer to Microsoft OCS |
ASTERISK-16308: cell phone get disconnected from asterisk |
ASTERISK-16309: [patch] Cannot join a queue when all members are paused and "joinempty=no". |
ASTERISK-16310: Application pickup doesn't work well |
ASTERISK-16311: ALSA Channel and MeetMe Conference |
ASTERISK-16312: [patch] file.o now needs to include directory DAHDI_INCLUDE |
ASTERISK-16313: AVAILSTATUS set to 20 |
ASTERISK-16314: [patch] Message on stale nonce give 'to' instead of 'from' |
ASTERISK-16315: [patch] remove extra line breaks from 'core show config mappings' |
ASTERISK-16316: [patch] enable autosystemname within asterisk.conf by default |
ASTERISK-16317: MeetMe time limited conference sound files not played |
ASTERISK-16318: noise |
ASTERISK-16319: [patch] crash if 'dahdi destroy channel' destroys a channel in a call |
ASTERISK-16320: [patch] applicationmap groups do not work (either lack of documentation, or feature incomplete) |
ASTERISK-16321: Asterisk LDAP Modify |
ASTERISK-16322: [patch] [regression] No CDR after originate from manager |
ASTERISK-16323: Parking a call calls out the parked extension on the parked channel - blind transfert on Polycom |
ASTERISK-16324: One-way audio after transfer |
ASTERISK-16325: [patch] Four Dahdi Events not recognized from chan_dahdi to disable Echo Cancelation |
ASTERISK-16326: Crash in sip_alloc |
ASTERISK-16327: AGI manager command is not consistent with other commands |
ASTERISK-16328: odbc can't insert specyfic record |
ASTERISK-16329: [patch] Add support for phones with less than 3 LCD lines |
ASTERISK-16330: No CDRs after AMI Redirect |
ASTERISK-16331: No ringback tone generated |
ASTERISK-16332: app_echo causing failures in dialplan? |
ASTERISK-16333: Conference with 60 Channels out of memory |
ASTERISK-16334: [patch] Func Strings Backported from Trunk |
ASTERISK-16335: [patch] Documentation about how sounds are generated |
ASTERISK-16336: Billsec is zero although disposition is Answered |
ASTERISK-16337: [patch] Deadlock is happended.( between channel and sip_pvt lock) |
ASTERISK-16338: Answer not working, maybe chan_oss problem, may be chan_sip problem. |
ASTERISK-16339: [patch] wrong SRV query for TLS connection |
ASTERISK-16340: [patch] After a blind transfer by the calling party the transferees peer cannot be dialed again within the same call |
ASTERISK-16341: [patch] IPv6 - Potential issue in via header parsing |
ASTERISK-16342: asterisk chrased while playing MP3 [WARNING[21708]: mp3/interface.c:215 decodeMP3: Junk at xxxxx] |
ASTERISK-16343: [patch] Configure sets ac_cv_fork_works=no incorrectly for uclibc |
ASTERISK-16344: [patch] main/netsock2.c fails to build do to missing constants |
ASTERISK-16345: [patch] convert 6 modules to use <module> show settings |
ASTERISK-16346: [patch] Chanspy hangs up the targetted phone call. |
ASTERISK-16347: [patch] sounds/Makefile ignores checksum errors |
ASTERISK-16348: [patch] STUN support not reliable |
ASTERISK-16349: [patch] DNID not cleared when channel hang up |
ASTERISK-16350: After placing a call on hold the MoH does not stop |
ASTERISK-16351: "I should never be called!" message printed from channels/chan_iax2.c |
ASTERISK-16352: [patch] Dialplan execution stops after awhile on an attended transfer by the calling party (with use of Dial 'g' option) |
ASTERISK-16353: One-way audio after exit from queue or after transfer |
ASTERISK-16354: voicemailpwcheck.py script for use with externpasscheck |
ASTERISK-16355: [patch] Chanspy Keeps using G729 Encoder licenses even after the spying channel hangs up. |
ASTERISK-16356: [patch] (Regression) Pickup from Grandstream BLF button ignores the context specified in Pickup command |
ASTERISK-16357: chan_mobile unable to connect to cellphone |
ASTERISK-16358: Manager GetVar on unset variable causes segfault |
ASTERISK-16359: [patch] STRFTIME in globals using commas returns incorrect value |
ASTERISK-16360: Segmentation fault when loading res_srtp.s on 64-Bit Debian |
ASTERISK-16361: Crash when hangup a srtp call |
ASTERISK-16362: Unhold issue |
ASTERISK-16363: [patch] reset visible_indication after call answering |
ASTERISK-16364: Asterisk crash if T.38 negotiation timed out |
ASTERISK-16365: [patch] dialplan reload deadlocks in ast_rdlock_contexts when calling ast_hint_state_changed |
ASTERISK-16366: funcs/func_env.c fails to compile because of missing definition for LLONG_MAX in <limits.h> |
ASTERISK-16367: chan_sip case 491 |
ASTERISK-16368: [patch] Packecable NCS 1.0 - DQOS Gate ID option malformed |
ASTERISK-16369: SIPPEER() with ldap and useragent |
ASTERISK-16370: [Patch] Invert the objectname in the AMI answer of the iaxpeers |
ASTERISK-16371: Video is not sended when using IAX2 to interconnect both servers |
ASTERISK-16372: [Patch] Increment the Asterisk Call Manager version |
ASTERISK-16373: [patch] Add CHANNEL(checkhangup) function |
ASTERISK-16374: [patch] Duplicate of 0016402 - dahdi show channels does not show an outgoing call - with patch |
ASTERISK-16375: [patch] configure.ac now breaks when running bootstrap.sh |
ASTERISK-16376: Record files with noise and/or audio cutting when using mixmonitor with medium call trafic on Asterisk. |
ASTERISK-16377: app_amd total analysis time error with SIP and silence suppression on |
ASTERISK-16378: call recording in attended transfer |
ASTERISK-16379: CODEX SPEEX 16K |
ASTERISK-16380: [patch] Using app_jack with JACK_HOOK will result in noise rather than the channel sound. |
ASTERISK-16381: Calling Party Category |
ASTERISK-16382: [patch] redirecting to IPv6 URIs |
ASTERISK-16383: [patch] IPv6: sip_uri_cmp |
ASTERISK-16384: [patch] IPv6: get_refer_info |
ASTERISK-16385: Ignoring SIPS requests |
ASTERISK-16386: [patch] Regression: Externip has port set to 0 |
ASTERISK-16387: Direct RTP failures |
ASTERISK-16388: [patch] update res_fax to the xml documentation style |
ASTERISK-16389: [patch] Reload logger with alternate configuration file |
ASTERISK-16390: ConfBridge crashes Asterisk |
ASTERISK-16391: Fetching Call with Grandstream-BLF (**${EXTEN}) |
ASTERISK-16392: Meetme PIN caching problem using realtime |
ASTERISK-16393: [patch] When using Local/ as members, language is not inherited |
ASTERISK-16394: [patch] Last pause information to queue members |
ASTERISK-16395: tcp-tls deadlock |
ASTERISK-16396: [patch] host not used in invite message, only the ip address. |
ASTERISK-16397: [regression][patch] Core dump when loading sip peers |
ASTERISK-16398: Fix select() usage in Asterisk |
ASTERISK-16399: load module error |
ASTERISK-16400: [patch] chan_skinny crashes asterisk when parking a call |
ASTERISK-16401: [patch] [branch] AppleRaisin - ASTDB over realtime |
ASTERISK-16402: [branch] SIP peer matching beyond proxy |
ASTERISK-16403: [patch] I could not find an easy way to access DAHDI channel and span number from the dialplan. |
ASTERISK-16404: [branch] Pinefrog - RTCP improvements |
ASTERISK-16405: Menuselect output no longer fits into 80x24 space |
ASTERISK-16406: Asterisk crashing in ast_readaudio_callback at file.c:762 |
ASTERISK-16407: Crashes due to memory issues in Manager.c [ _ast_malloc ] [Possibly related to 0017234] |
ASTERISK-16408: [patch] GCC 4.2.x optimizations result in improper behavior of GSM codec |
ASTERISK-16409: FAXOPT(modem)=V27,V29 does not have an effect on spandsp, fax still get received with V17 |
ASTERISK-16410: attended transfer |
ASTERISK-16411: [patch] subchannel remains half-open after call transfer |
ASTERISK-16412: [patch] [regression]context value from chan_dahdi.conf not used. |
ASTERISK-16413: [patch] Additional Makefile cleanup |
ASTERISK-16414: meetme |
ASTERISK-16415: [patch] Workaround required for environments not supporting locale |
ASTERISK-16416: CTRL-C from asterisk -vvvvvvvvc has strange results |
ASTERISK-16417: user get unreachable after some minutes, deadlock in ao2_lock |
ASTERISK-16418: [patch] When matching peers check invite from domain against domain list |
ASTERISK-16419: meetme |
ASTERISK-16420: " > doing dnsmgr_lookup" message appears on the console often |
ASTERISK-16421: AST_OPTION_ONLY tests wrong variable |
ASTERISK-16422: [patch] Asterisk 1.8-beta1 crash on "logger reoad" |
ASTERISK-16423: seg fault on reload from FreePBX 2.8 |
ASTERISK-16424: [regression] T.38 using linksys ATA no longer works with 1.8.0-beta1 (working with 1.6.2.10) |
ASTERISK-16425: [patch] Upgrading from 1.6.2.10 to 1.8-beta1 did not work with the original modules.conf |
ASTERISK-16426: [patch] HTTP always enabled, regardless of http.conf settings. |
ASTERISK-16427: [patch] Formats should have a load priority |
ASTERISK-16428: bad magic number : loops indefinitly |
ASTERISK-16429: [patch] sip/reqresp_parser.c:sip_uri_cmp_test failure |
ASTERISK-16430: TOS_SIP does not get set |
ASTERISK-16431: [patch] Add FIELDNUM() function, returns position of a field in a list |
ASTERISK-16432: CLI command reload crashes asterisk |
ASTERISK-16433: [patch] cli cmd 'fax set debug on' has no effect. Dont get any spandsp debug |
ASTERISK-16434: [patch] [regression] 1.6.2.10 sounds Makefile error prevents install in Centos 4.8 (x86) with GNU Make 3.80 |
ASTERISK-16435: [patch] dynamic_exclude_static option results in ACL errors |
ASTERISK-16436: SIP packets not marked with configured QoS setting |
ASTERISK-16437: [patch] Autoconf issues PGSQL_INCLUDE, etc. |
ASTERISK-16438: [patch] suggested documentation update to doc/backtrace.txt |
ASTERISK-16439: [patch] sounds/Makefile symbolic link has incorrect path with Make 3.80 |
ASTERISK-16440: Issue with transfers in chan_skinny |
ASTERISK-16441: [patch] res_musiconhold does not use generic timing interfaces |
ASTERISK-16442: Bridge with channel fails; channel format 0x0 (nothing) |
ASTERISK-16443: Not more than 256 character can be returned from column data |
ASTERISK-16444: [patch] chan_iax2 produces PeerState registered event without Address and Port |
ASTERISK-16445: Update chan_sip to use the security framework API |
ASTERISK-16446: [patch] chan_usbradio.c fails to build with --enable-dev-mode |
ASTERISK-16447: CDR(userfield) is not saved when called by Set(CDR(userfield)=...) activated by features.conf |
ASTERISK-16448: Bridge with sip channel fails; channel format 0x0 (nothing) |
ASTERISK-16449: [patch] Video RTP type in response SDP not matching the one in INVITE |
ASTERISK-16450: Only one certificate, multiple domains |
ASTERISK-16451: [patch] sip.conf lacks documentation for IPv6 |
ASTERISK-16452: [patch] reference leak when adding dynamic queue members |
ASTERISK-16453: Asterisk crashes with a segmentation fault in timing.c [ast_timer_ack] |
ASTERISK-16454: Incoming calls go straight to voicemail |
ASTERISK-16455: [patch] sip_poke_noanswer launch ast_devstate_changed everytime even a peer is still unreachable |
ASTERISK-16456: [patch] ast_sched_runq runs to much events if one event runs too long |
ASTERISK-16457: [patch] Asterisk 1.8 beta2 crash with configured MGCP/NCS when option wcardep = aaln/* is used |
ASTERISK-16458: requirecalltoken=no crashes Asterisk |
ASTERISK-16459: can not detect status of clients |
ASTERISK-16460: [patch] Asterisk crashes every less than 1 hours (arrount 20 mn) when using manager and http manager using asterisk-1.4.35-rc1 |
ASTERISK-16461: queue shows the wrong agent in queue_log (CONNECT event) when the call is pickup up by another phone |
ASTERISK-16462: Some hanging ReceiveFAX app after a dax y with 1000 Fax |
ASTERISK-16463: Wrong DIALSTATUS/HANGUPCAUSE set on receiving 416 reply to INVITE |
ASTERISK-16464: [patch] Error in sig_pri.c issues compiler warning and causes chan_dahdi to be unloadable - 1.8.0-beta2 |
ASTERISK-16465: [PATCH] Add Q.850 reason headers to SIP CANCEL messages. |
ASTERISK-16466: context from chan_dahdi.conf is cutted in pbx extensions support, callerid also wrong |
ASTERISK-16467: [patch] SIP channel AMI session timeout events feature |
ASTERISK-16468: [patch] DTMF CallerID failing wirh dtmfcidlevel with high value |
ASTERISK-16469: What is ccss.conf ? |
ASTERISK-16470: Turn on ALWAYSAUTHREJECT by default |
ASTERISK-16471: [patch] Outbound proxy set to IPv6 address in square brackets doesn't work |
ASTERISK-16472: [patch] Asterisk just reads the first "Accept" header |
ASTERISK-16473: [patch] Format change suggestion for output of 'sip show peers' |
ASTERISK-16474: CDR(userfield) is not saved when called by Set(CDR(userfield)=...) activated by features.conf |
ASTERISK-16475: CDR user fields not updated and CDR() returns invalid data when using Queue with "c" flag |
ASTERISK-16476: WARNING[7623]: translate.c:204 framein: g729tolin did not upte samples 0 |
ASTERISK-16477: Messagedelivery on Exchange 2003 via imap storage |
ASTERISK-16478: Wrong Messagedelivery on Exchange 2003 |
ASTERISK-16479: [patch] Strange booting behavior depending on nice value |
ASTERISK-16480: pressing 0 in voicemail spawns extension (context, o, 0) |
ASTERISK-16481: Create doxygen page on asterisk.org |
ASTERISK-16482: [patch] Asterisk is restarted to make a single ring |
ASTERISK-16483: AudioHook Inherit doesn't work after atxfer. |
ASTERISK-16484: [patch] Cleanup: consolidate call answering |
ASTERISK-16485: call overlapping |
ASTERISK-16486: [patch] SIP peers memory leak |
ASTERISK-16487: [patch] HTTP redirect support for calendars |
ASTERISK-16488: doing dnsmgr_lookup stops working |
ASTERISK-16489: [patch] add imapserver in voicemail.c |
ASTERISK-16490: [patch] tcptls.c:350 Unable to connect SIP socket Connection refused |
ASTERISK-16491: chan_misdn fails to build |
ASTERISK-16492: Incorrect hangupcause (says 34) |
ASTERISK-16493: [patch] If EWS request fails, asterisk crashes because of double free |
ASTERISK-16494: [patch] No support for Cisco 7906 handset |
ASTERISK-16495: [patch] Encoded URI in a subscription does not work |
ASTERISK-16496: [patch] Events are visible after they were removed from EWS calendar |
ASTERISK-16497: [regression] chan_local is hanging up the call upon first channel answer when calling from asterisk manager (inconsistent) |
ASTERISK-16498: Update asterisk.org to have proper merge order documentation |
ASTERISK-16499: [patch] Asterisk crashes at startup on Solaris SPARC |
ASTERISK-16500: [patch] Missing semicolon in SIP-Notify |
ASTERISK-16501: [patch] use a realtime __options__ column to store several name=value pairs |
ASTERISK-16502: sip.conf register in realtime MySQL DB |
ASTERISK-16503: chan_mgcp does not send CRCX with SDP |
ASTERISK-16504: [patch] segfault on dialplan reload |
ASTERISK-16505: using chan_iax2 ringing received from destination sip channel doesn't always get propogated to originating sip |
ASTERISK-16506: random Crash |
ASTERISK-16507: [patch] Allow chan_sip's support for INFO/Record request to also use dynamic features |
ASTERISK-16508: [patch] SIP autocreate peers disappear on "sip reload" |
ASTERISK-16509: Another sip_peer structure leak |
ASTERISK-16510: ReceiveFax doesn't work with attended transfer |
ASTERISK-16511: core dump in ast_cdr_getvar |
ASTERISK-16512: [patch] ERROR[7169] astobj2.c: bad magic number 0x0 for 0x8b1c3d0 |
ASTERISK-16513: Direct Media mode - Wrong Coder Selected |
ASTERISK-16514: [patch] [regression] Forwarding a voicemail with a prepended message does not work |
ASTERISK-16515: ACCEPT message should contain an FORMAT2 ie as well as a FORMAT ie |
ASTERISK-16516: SIP REFER auth fails, RTP timeout ignored, and other discrepancies between e/ingress calls |
ASTERISK-16517: hangup when using parkandannouce |
ASTERISK-16518: Music on hold doesn't recover very cleanly when it can't play a file |
ASTERISK-16519: [patch] Function CONNECTEDLINE causes Asterisk to exit |
ASTERISK-16520: T.38 fax sending succesful but asterisk reporting failure |
ASTERISK-16521: Lock during module reload |
ASTERISK-16522: [patch] Added CURL() Function Timeout Argument |
ASTERISK-16523: [patch] Cleanup: consolidate offhook (new call) |
ASTERISK-16524: Segmentation fault in scheduled event |
ASTERISK-16525: AGI Hangup does not hangup channel on FastAGI |
ASTERISK-16526: Peer Qualify is not Working |
ASTERISK-16527: Asterisk handles 182 Queued event as 180 Ringing |
ASTERISK-16528: Enable to use '-' sign in CallerID |
ASTERISK-16529: Fix for possible memory leak |
ASTERISK-16530: Turn on mode=new as default (say.conf) |
ASTERISK-16531: [patch] AGENTBYCALLERID not set for agent logged from AMI |
ASTERISK-16532: asterisk es-gsm sounds url goes timeout |
ASTERISK-16533: dun query EID allways return "DUNDi Query EID returned no results" |
ASTERISK-16534: [patch] A 'Directed pickup' call is later unable to Hold or Transfer |
ASTERISK-16535: [regression] After 1.6.2.8 the #include directive does not work if missing closing double quote |
ASTERISK-16536: [patch] certain exception states do not jump to 'e' error extension properly |
ASTERISK-16537: No CDR with originate from manager and then an redirect to a dial from manager |
ASTERISK-16538: Unavailable presentation number status is ignored if sendrpid is true (chan_sip) |
ASTERISK-16539: Dropped calls when entering meetme conference |
ASTERISK-16540: [patch] Registration of SIP phone denied on transport=unknown |
ASTERISK-16541: saydigit not working |
ASTERISK-16542: [patch] IPv6: SIp show settings doesn't show dual stack support |
ASTERISK-16543: [patch] SIP domains automatically add 0.0.0.0 and :: for IPv6 |
ASTERISK-16544: [patch] say.conf has problem with large numbers |
ASTERISK-16545: [patch] IPv6: System configured for only IPv4 tries sending to IPv6 |
ASTERISK-16546: [patch] say.conf dont have the same amount of rule's as say.c |
ASTERISK-16547: [patch] say.conf added support for Danish |
ASTERISK-16548: [patch] Support for priorities and categories according to RFC 5545 |
ASTERISK-16549: Menuselect space invaders does not permit screen resizing during the game |
ASTERISK-16550: IPv6: Asterisk tries to do DNS resolution even though outbound proxy is configured |
ASTERISK-16551: sip show settings: Internal IP with bindaddr=:: |
ASTERISK-16552: Process use of 100% |
ASTERISK-16553: T.38 Re-Invite - Asterisk Sends Call to Incoming Context of Extensions.conf |
ASTERISK-16554: DTMF Manager events missing with some codecs on bridged SIP calls |
ASTERISK-16555: https://issues.asterisk.org/view.php?id=17801 |
ASTERISK-16556: DTMF ISSUE |
ASTERISK-16557: Fail to transfer a call after putting that call to parking lot, taking it on the parking lot, and try to transfer (atxfer) |
ASTERISK-16558: Asterisk does not properly align SDP "m=" lines when answering an SDP offer (provoking a T.38 negociation issue) |
ASTERISK-16559: Queue Stats being reset by app_queue.so reload |
ASTERISK-16560: SIp/SDP for Non-NAT phone not used during 183 Session Progress with Non-NAT peer |
ASTERISK-16561: Error destroying mutex &thread->lock: Device or resource busy |
ASTERISK-16562: Asterisk with SRTP Record and Playback Issue |
ASTERISK-16563: Defining different moh-classes, but always using "default" |
ASTERISK-16564: allowguest should be off by default in sip.conf |
ASTERISK-16565: SIP INVITE |
ASTERISK-16566: [patch] trustrpid and sendrpid global settings not in "sip show settings" |
ASTERISK-16567: [regression] X-Lite disconnects because of RTCP timeout (see also #17236 and #17559) |
ASTERISK-16568: [patch] Manager interface requires a space after the : |
ASTERISK-16569: configure asterisk with ffmpeg for use console_video |
ASTERISK-16570: Usage of unavailable channels on multi-PRI NT spans (NFAS) |
ASTERISK-16571: Special Meetme Plan |
ASTERISK-16572: chanspy cuts the voice between caller and callee if Local channel is used |
ASTERISK-16573: [patch] If "notifycid=ignore-context" is set in sip.conf Asterisk core dump. |
ASTERISK-16574: [patch] Unable to build main/netsock.c on Nexenta CP3 |
ASTERISK-16575: [patch] greetingsfolder variable should be greetingfolder |
ASTERISK-16576: [patch] IMAP with maildir formats do not handle greetings correctly |
ASTERISK-16577: update_odbc always reports that : Key field 'ipaddr' does not exist in table ... Update will fail |
ASTERISK-16578: Record audio channels gets out of sync |
ASTERISK-16579: [patch] Q931 - Sending PROGRESS after sending ALERTING is a protocol error |
ASTERISK-16580: problem with the voicemail button (msgs) |
ASTERISK-16581: [patch] the display does not properly display the caller when a call is established. - Cisco 7910 |
ASTERISK-16582: [patch] Merging events for Exchange web service doesn't work as expected, resulting in only one event in calendar |
ASTERISK-16583: [patch] chan_sip fails to remove hold when receving a reINVITE without SDP |
ASTERISK-16584: VoiceMailMain with the application restarts asterisk. |
ASTERISK-16585: Remove dahdi and meetme dependency for page function |
ASTERISK-16586: Video RTP is not sended to originating SIP extension when using IAX2 to interconnect both servers |
ASTERISK-16587: Crash in ast_frame_free |
ASTERISK-16588: ast_odbc_direct_execute: SQL Exec Direct failed while storing CDR rto Oracle 11g |
ASTERISK-16589: Create IMAP driver for ast_storage |
ASTERISK-16590: asterisk 1.4.3x codec packetization issue |
ASTERISK-16591: Sip Reason header "Call completed elsewhere" is not passed through local channel in queues |
ASTERISK-16592: [patch] When you press the redial button restarts asterisk. |
ASTERISK-16593: ast_add_hint deadlock in pbx.c MUTEX ast_hint_state_changed |
ASTERISK-16594: Can't compile |
ASTERISK-16595: [regression][patch] RTP Packets Not Set With QOS |
ASTERISK-16596: Possible memory leak in originate |
ASTERISK-16597: [patch] contact header does not get ast_uri_encoded value from p->exten, but dialplan does |
ASTERISK-16598: [patch] 'dialplan save' crash when '|' (pipe) is used. Patch included. |
ASTERISK-16599: Asterisk does not recognize RTP Payload 126 |
ASTERISK-16600: res_stun_monitor prevents asterisk from loading if it does not have a config file. |
ASTERISK-16601: chan_multicast_rtp.so MulticastRTP no audio when using Page() App |
ASTERISK-16602: [patch] App Dial does not respect TON and fails when etsi call rerouting is used |
ASTERISK-16603: say_date_generic in app_playback.c ignores timezonename |
ASTERISK-16604: [patch] Fake answer start in app_dial |
ASTERISK-16605: [patch] empty CDR variables and everything that goes after is not shown |
ASTERISK-16606: Asterisk is restarted by pressing the button "MSGS" during a call. |
ASTERISK-16607: [patch] Asterisk 1.8.0-beta3 DNSMGR address corruption |
ASTERISK-16608: [patch] Call file errors in Asterisk 1.8beta |
ASTERISK-16609: Screech Sound (loud noise) heard every 100-200 calls in call centre environment |
ASTERISK-16610: problem to reload the module skinny when active calls. |
ASTERISK-16611: [patch] New Manager Action: MeetmeListRooms |
ASTERISK-16612: [patch] Cleanup: consolidate new call |
ASTERISK-16613: [patch] MeetMe PIN handling broken |
ASTERISK-16614: [patch] Caller presentation in the Queue |
ASTERISK-16615: Debian init script does not work |
ASTERISK-16616: ODBC connection to mssql does not work |
ASTERISK-16617: ao2_t_callback of dialog_needdestroy is called every loop of do_monitor |
ASTERISK-16618: Unable to use IPv4 addresses for a TCP host when using IPv6 |
ASTERISK-16619: Blackberry 9000 does not reconnect when enters the area |
ASTERISK-16620: Segmentation Fault: bridge_softmix.c Line 170 of softmix_bridge_write |
ASTERISK-16621: [patch] Asterisk not honouring umask when creating vm directories |
ASTERISK-16622: Reloads of manager.conf do not properly handle the resetting of options |
ASTERISK-16623: No announce doesn't get reloaded in queue configuration |
ASTERISK-16624: [patch] schedule_delivery calls ast_bridged_channel() on an unlocked channel |
ASTERISK-16625: schedule_delivery calls ast_bridged_channel() on an unlocked channel |
ASTERISK-16626: DTMF not logged to console when configured in logger.conf |
ASTERISK-16627: Segfault in iax2_process_thread |
ASTERISK-16628: SRTP stops working anymore beta4 |
ASTERISK-16629: [patch] AST_MAX_EXTENSION limitation on hint string length |
ASTERISK-16630: digium S400P Card dose not show FXS Moduls with asteisk 1.6 and higher |
ASTERISK-16631: Random Deadlocks maybe in find_call() of chan_sip. or ast_call() of channel.c |
ASTERISK-16632: Enhance core APIs to support T.38 Gateway design |
ASTERISK-16633: [patch] downsampling slinear16 to slinear (or ulaw or alaw or g729) results in truncation |
ASTERISK-16634: [patch] IAXregistry AMI does not return ActionID data |
ASTERISK-16635: Crashed after dial local without /n |
ASTERISK-16636: USERUSERINFO not documented in channelvariables.txt |
ASTERISK-16637: sendrpid and trustrpid support from users.conf |
ASTERISK-16638: RealTime doesn't set regserver value in mysql |
ASTERISK-16639: call-limit is not removed by "sip reload" |
ASTERISK-16640: ConfBridge crashes when leave simultaneously |
ASTERISK-16641: Debug message when Asterisk loads |
ASTERISK-16642: [patch] Minor fixes to multiparking |
ASTERISK-16643: [patch] Allow user more control over parked calls that expire from a one touch park |
ASTERISK-16644: nonlatin strings are not retrieved |
ASTERISK-16645: [patch] Mutex 'dialog' freed more times than we've locked ! and Error releasing mutex. |
ASTERISK-16646: iterate through all ao2_sip dialogs on every subscribe is slow, and not needed |
ASTERISK-16647: Asterisk Crashes after 70 calls |
ASTERISK-16648: INVITE with Replaces: breaks Monitor() call recording. |
ASTERISK-16649: [patch] Peer does not hang up when caller hangup while app_dial is executing - Deadagi |
ASTERISK-16650: Chanspy hangs up at least one of the spyed channels, when starting to spy |
ASTERISK-16651: [patch] atxfer broken on 1.6.2.11 |
ASTERISK-16652: [patch] merging categories from with static config to have realtime feature for register |
ASTERISK-16653: [patch] debian warnings on make config |
ASTERISK-16654: [patch] SIP peer wrong URI an to: tag |
ASTERISK-16655: Create commercial/external modules for Asterisk 1.8.0 |
ASTERISK-16656: [patch] Crash with autoclear=yes with MYSQL() application |
ASTERISK-16657: SET CALLERPRES() = prohib not working on pri with |
ASTERISK-16658: [patch] [regression] T.38 only invites (Fax Only Calls) are no longer possible since Asterisk 1.4.25 |
ASTERISK-16659: sip peer isnt freed |
ASTERISK-16660: [patch] res_agi.c:handle_getvariablefull() failes to unlock channel if given channel name of the channel that AGI is running on |
ASTERISK-16661: [patch] No audio is heard when calling from asterisk to Cisco CallManager. |
ASTERISK-16662: Asterisk messes up progress indication on q.931 ALERTING message |
ASTERISK-16663: VERBOSE message shows up on console when 'debug' enabled in logger.conf |
ASTERISK-16664: strndup doesn't exist on mac os/X |
ASTERISK-16665: Asterisk Crash on RTCP package in SRTP mode |
ASTERISK-16666: [patch] Negative filter values |
ASTERISK-16667: [patch] Wrong URI send if P-Assterted-Identiy is sent and caller is anonymous -> leads to reject on Aastra phone |
ASTERISK-16668: ACK tone interupted - Jitterbuffers do not function properly as AlarmReceiver App does not send RTP regularly |
ASTERISK-16669: lack of svn:ignore for files in "tests" directory |
ASTERISK-16670: Asterisk Crash |
ASTERISK-16671: New calls generate ambiguous message on the Asterisk console |
ASTERISK-16672: high chan_sip use count (mem leak) |
ASTERISK-16673: [regression] Newchannel event is missing during masquerading process |
ASTERISK-16674: Random deadlock in res_config_mysql |
ASTERISK-16675: [patch] Application SayUnixTime always jumps to extension when digit sent |
ASTERISK-16676: DAHDIRAS fails to properly initiate pppd unless asterisk is running as root |
ASTERISK-16677: [patch] Crash, possibly with MYSQL() or with GotoIfTime() inside a Macro() |
ASTERISK-16678: Asterisk crash when bridging and masquerading channels |
ASTERISK-16679: [patch] crash in __ast_manager_event_multichan |
ASTERISK-16680: No audio when use app_festival |
ASTERISK-16681: Crash on "voicemail show users for default" |
ASTERISK-16682: gtalk Unable to allocate RTP socket |
ASTERISK-16683: [patch] chan_sip send MWI Notify, even if there is no app_voicemail loaded |
ASTERISK-16684: Issues with DTMF triggered attended transfers |
ASTERISK-16685: Qwell is a nub. |
ASTERISK-16686: [patch] GET DATA problem with pipes |
ASTERISK-16687: PBX freeze |
ASTERISK-16688: H323 Remote IP |
ASTERISK-16689: chan_dahdi does not detect polarity change when using AEX800P/wctdm24xxp |
ASTERISK-16690: Meetme unreliable functionality |
ASTERISK-16691: 'confbridge list' and 'confbridge kick' CLI commands |
ASTERISK-16692: deadlock in local_bridgedchannel during masquerade |
ASTERISK-16693: [patch] [patch/branch] Add CLI "cdr pgsql status" |
ASTERISK-16694: [patch] ACK tone not reliable on embedded platform with low CPU power |
ASTERISK-16695: [patch] PickupChan() is not working with full channel name |
ASTERISK-16696: [patch] constantssrc option for h323 |
ASTERISK-16697: [patch] segfault on voice frame handling |
ASTERISK-16698: large memory consumption of udptl.c module |
ASTERISK-16699: Asterisk random crashed 4 times one day |
ASTERISK-16700: SIP Trunk ${DIALSTATUS} wrong return code - it is always "ANSWER" status |
ASTERISK-16701: [patch] asterisk could not register to asterisk with pedantic=yes |
ASTERISK-16702: [patch] Hints for non-existent devices are in an Idle state |
ASTERISK-16703: [patch] chan_iax2 - timing interface missing |
ASTERISK-16704: Realtime SIP Registration lost when performing a SIP RELOAD |
ASTERISK-16705: T.38 Fax Fallback to Pass-Through didn't work |
ASTERISK-16706: chan_skype generate kernel error when starts asterisk |
ASTERISK-16707: [patch] func_strings would like a STRREPLACE function |
ASTERISK-16708: Strange Message in 1.6.2 (64 Bits) only |
ASTERISK-16709: Status of SIP Agent with call-limit 0 never changes |
ASTERISK-16710: memory leak in chan_sip.c |
ASTERISK-16711: [patch] Exceptionally long queue length queuing to XXXXX |
ASTERISK-16712: Asterisk does not re-read res_fax.conf upon reload |
ASTERISK-16713: potential channel/queue deadlock when setqueuevar enabled |
ASTERISK-16714: appdata arguments receive trailing slash when passed to Realtime |
ASTERISK-16715: realtime appdata argument separator still '|' |
ASTERISK-16716: Asterisk crashes on local_ast_moh_start (Segmentation Fault) |
ASTERISK-16717: Incoming SIP UDP packets are ignored. |
ASTERISK-16718: [patch] func_uri could use a QSFIELD function to parse x-www-form-urlencoded data |
ASTERISK-16719: XMPP PubSub Distributed Device State don't work? |
ASTERISK-16720: Small extconfig/CLI change |
ASTERISK-16721: The manager interface listens even when it is not enabled |
ASTERISK-16722: Crash when assigning 2 return vallues to an ARRAY with FUNC_ODBC call |
ASTERISK-16723: [patch] inefficient strlen() in ast_uri_decode loop |
ASTERISK-16724: Crash on load |
ASTERISK-16725: [patch] Crash in check_sounds |
ASTERISK-16726: Unable to have videosupport with trunk=yes on an IAX chan |
ASTERISK-16727: [patch] func_curl CURLOPT lacks a querycomponent (+-decoding) hashcompat mode |
ASTERISK-16728: Incorrect registrations and/or Check Authorization of NAT SIP devices |
ASTERISK-16729: Chanspy using spy groups not getting audio from calls placed to a queue using local channel |
ASTERISK-16730: SIP brute force attemps having a DoS effect |
ASTERISK-16731: Asterisk freezes when reloading dialplan |
ASTERISK-16732: report |
ASTERISK-16733: SIP Channel config will not reload if Name is only different by CaSe |
ASTERISK-16734: Outgoing custom registration length always reset to defaultexpiry |
ASTERISK-16735: bad dialog-info remote information |
ASTERISK-16736: SIP session timers do not transmit session expiry or refresher timer in 200 OK |
ASTERISK-16737: [patch] func_string FILTER contains an infinite loop |
ASTERISK-16738: Unsolicitated MWI to co-exist with Exchange 2010 |
ASTERISK-16739: chan_sip doesn't get built if missing OpenSSL dependencies |
ASTERISK-16740: Crash and core dump on dial |
ASTERISK-16741: Bad assumption in ast_ouraddrfor in chan_sip regarding extern IP and IPv6 |
ASTERISK-16742: sometimes crash after Unable to write to alert pipe |
ASTERISK-16743: SRTP does not work if used togather with TLS |
ASTERISK-16744: Problems with loading of res_calendar and its sister modules |
ASTERISK-16745: [patch] New variable VMFILENAME with the name of the new record file |
ASTERISK-16746: Register an SIP peer through CLI |
ASTERISK-16747: [regression] 'D' option of MeetMe does not work for 1st caller (PIN not asked) |
ASTERISK-16748: [patch] res_config_pgsql needs db reconnection support |
ASTERISK-16749: RTPAUDIOQOSJITTERBRIDGED and other variables contains nothing if call is terminated from DAHDI |
ASTERISK-16750: ICC compilation bug |
ASTERISK-16751: macro does not match Goto extension s-NOANSWER when extenpatternmatchnew=yes |
ASTERISK-16752: [patch] failure to cross compile asterisk-1.8.0-rc3 |
ASTERISK-16753: [patch] Asterisk fails to recognize SUBSCRIBE retransmissions and tries to re-authenticate them, which breaks presence on polyco |
ASTERISK-16754: Cannot compile with "make menuselect" |
ASTERISK-16755: Call hangs on causecode 100 |
ASTERISK-16756: Wrapuptime sometimes not respected |
ASTERISK-16757: SIP 200 OK withou ACK |
ASTERISK-16758: can't register to different sip accounts when registered to same IP/DNS |
ASTERISK-16759: Several issues with app_meetme (realtime) after upgrading from 1.6.1.20 to 1.6.2.13 |
ASTERISK-16760: DTMF tones fail DAHDI>DAHDI |
ASTERISK-16761: Segmentation fault caused by "core restart when convenient" while SRTP call is active |
ASTERISK-16762: differences between meetme error handling |
ASTERISK-16763: [regression] Caller ID in SIP PAI Header is anonymous instead of Number |
ASTERISK-16764: [patch] Call files generate two warning logs after each successful completion |
ASTERISK-16765: app_fax.c: Transmission error for successfully transmitted faxes |
ASTERISK-16766: [patch] dialgroup fills database with duplicates |
ASTERISK-16767: [patch] change Language channel but still english voice prompts playback |
ASTERISK-16768: Memory leak happens when repeatedly tested |
ASTERISK-16769: Asterisk always crashes when linked against glibc 2.12.1 |
ASTERISK-16770: Asterisk refuses to accept INVITE although auth info is correct |
ASTERISK-16771: unexpected output from command line, when issuing commands from shell |
ASTERISK-16772: Many "rtp.c: RTP Read too short" errors followed by memory exhaustions and crash |
ASTERISK-16773: [patch] tos_sip and tos_audio doesn't work on IPV6 |
ASTERISK-16774: Dial with use of feature options corrupts rfc2388 DTMF durations |
ASTERISK-16775: Manager class log is not used |
ASTERISK-16776: asterisk stops too late |
ASTERISK-16777: several filename bugs in Record() application |
ASTERISK-16778: Playback of audio file (gsm,ulaw) audio and dialplan stops when caller is inbound over iax2 |
ASTERISK-16779: Cannot disallow unknown format '' |
ASTERISK-16780: [patch] Add Undocumented Variables to phoneprov.conf.sample |
ASTERISK-16781: [patch] substitute_escapes() not used on per-mailbox 'emailbody' variable |
ASTERISK-16782: [patch] app_ivonacl.c text-to-speech application |
ASTERISK-16783: Playback stalls when playing demo-congrats to an IAX2 channel |
ASTERISK-16784: [patch] Message lost when sox fails to re-encode with 'volgain' |
ASTERISK-16785: [patch] Report what extension called a failed macro |
ASTERISK-16786: looped local channels, some not optimize out. |
ASTERISK-16787: random chan_sip deadlock in ast_sched_add_variable on reload |
ASTERISK-16788: Voicemail don't work on Asterisk 1.8.0 RC3 |
ASTERISK-16789: [patch] [regression] faxdetect now must apply to all dahdi channels; change from 1.6.2.x |
ASTERISK-16790: AEL inconsistency with global vs. local variable de-quoting |
ASTERISK-16791: 1.4 Passthru Receives T38 Re-Invite and passes on a non-T38 invite to other peer. |
ASTERISK-16792: Reregistration with small (20 sec) Expiry fails |
ASTERISK-16793: Ability to escape # in spool files |
ASTERISK-16794: Early bind of UDPTL ports can create a DoS condition |
ASTERISK-16795: [patch] Allow emailsubject and emailbody to be specified in voicemail.conf |
ASTERISK-16796: rtautoclear = no or rtautoclear = 0 not respected in sip.conf |
ASTERISK-16797: Non-existent include in extensions.conf halts processing of conf file |
ASTERISK-16798: [patch] chan_sip loses port information for peers in memory when using bindaddr=:: |
ASTERISK-16799: Callee declined when 'beep' audio file does not exist |
ASTERISK-16800: [patch] stuck channels if followme context doesnt exists |
ASTERISK-16801: Session timer refresher role swap on reinvite requested, but not not honoured |
ASTERISK-16802: [patch] application not properly unregister in voicemail |
ASTERISK-16803: [patch] Oneway audio from SIP phone to FXS port after FXS port gets a CallWaiting pip |
ASTERISK-16804: [regression] DTMF on agent channel causes high CPU |
ASTERISK-16805: Remote-Party-ID instead of RFC 4916 (From change)? |
ASTERISK-16806: SIp channel stop work |
ASTERISK-16807: Unable to use G.729 codec with DUNDI |
ASTERISK-16808: Missing error message from System() app, if sh binary is missing |
ASTERISK-16809: [patch] cannot override mailbox/context per user in users.conf. |
ASTERISK-16810: Can't compile cdr_mysql addon |
ASTERISK-16811: SRTP enable disable from dialplan |
ASTERISK-16812: [patch] AMI command 'MeetMeList' responds with error when no conferences exist |
ASTERISK-16813: [regression] DeadAGI sends signal and closes AGI I/O when a caller hangs up. |
ASTERISK-16814: Passing multiple arguments to applications fails |
ASTERISK-16815: [patch] segfault when performing "module reload res_odbc.so" |
ASTERISK-16816: [patch] Load tests with sipp against asterisk |
ASTERISK-16817: ast_app_dtget inconsistency |
ASTERISK-16818: Meetme + DTMF = Problem |
ASTERISK-16819: Ringing / Alerting Problem |
ASTERISK-16820: The 2nd time you dial a non-exist jabber peer, Asterisk will crash |
ASTERISK-16821: SLIN/SLIN16 translation path is not optimal |
ASTERISK-16822: Channel Variable SMSSRC not set properly |
ASTERISK-16823: asterisk -rnx causes many errors on broken pipe |
ASTERISK-16824: Music on hold not released after answer of bridged call |
ASTERISK-16825: conflicting types for 'timersub' |
ASTERISK-16826: Wideband Streaming Music-On-Hold (slin16) |
ASTERISK-16827: [patch] Add ConnectedLineNum/Name headers to output of manager action Status |
ASTERISK-16828: Segfault related to realtime peers and the astDB |
ASTERISK-16829: Music on hold sometimes works, sometimes doesn't |
ASTERISK-16830: [patch] crashing func_curl hashcompat with invalid data |
ASTERISK-16831: [patch] deny=all in iax.conf results in segmentation fault |
ASTERISK-16832: T.38 only negotiate 2400 Baud for Faxes |
ASTERISK-16833: res_musiconhold does not compile |
ASTERISK-16834: [patch] hint state changes deadlock problem |
ASTERISK-16835: Segfault when shutting down and ongoing traffic to ConfBridge application |
ASTERISK-16836: [patch] A call locks when sending SIP originate from AMI script |
ASTERISK-16837: [patch] Duplicated event on AMI interface |
ASTERISK-16838: Channel redirect doesn't work |
ASTERISK-16839: Peer goes unreachable when placing a call from it. |
ASTERISK-16840: EAGI application should support wideband audio |
ASTERISK-16841: Application ICES should support wideband audio |
ASTERISK-16842: [patch] record priv-recordintro as sln, not gsm |
ASTERISK-16843: Certain Unicode causes res_jabber to report "Parsing failure: Invalid XML" |
ASTERISK-16844: Asterisk manager redirect results no cdr |
ASTERISK-16845: [patch] useropts not initialized |
ASTERISK-16846: [patch] [branch] IAX2 on Realtime doesn't resolv DNS lookups |
ASTERISK-16847: Blind transfer failure, A calls B, B transfers to C |
ASTERISK-16848: [patch] Malformed XML response |
ASTERISK-16849: Indicate SRTP + Feature reqest |
ASTERISK-16850: sip_setoption: Unknown option: 9 |
ASTERISK-16851: RFC2833 DTMF generation broken due to SSRC change on bridges channels |
ASTERISK-16852: Fax For Asterisk - T2_TIMEOUTand REMOTE_DISCONNECT errors |
ASTERISK-16853: Call dropped on SIP REFER in 1.8.0 |
ASTERISK-16854: [patch] roundf causing asterisk to fail to compile |
ASTERISK-16855: [patch] Loading chan_iax2.so allocates >100M RSS |
ASTERISK-16856: FaxLicense event is missing ActionID |
ASTERISK-16857: FaxLicenseList complete event is missing the trailing newline |
ASTERISK-16858: Response to FaxLicenseStatus is missing the trailing newline |
ASTERISK-16859: Jabber service unable to connect to Zimbra XMPP server |
ASTERISK-16860: Not responding to 401 request for authentication when placing outgoing call |
ASTERISK-16861: chan_sip hangs |
ASTERISK-16862: Exchange UM and SIP 302 sets Diversion on INVITE |
ASTERISK-16863: [patch] IAX2 registration issues |
ASTERISK-16864: When i call from Ekiga to my nortel phone asterisk crashes |
ASTERISK-16865: deadlock on 1.8.0-rc2 and crash on 1.8.0 with multipule sip channels |
ASTERISK-16866: PBX hangup |
ASTERISK-16867: [patch] Hold/Unhold not possible with SIP using SRTP |
ASTERISK-16868: [patch] Function LOCK doesn't wait for the lock as documented |
ASTERISK-16869: Asterisk crash using PickUp Application |
ASTERISK-16870: [patch] Support "mailbox" setting in users.conf |
ASTERISK-16871: [regression] ast_md5_hash sometimes create incorrect md5 digest |
ASTERISK-16872: Deadlock in chan_sip |
ASTERISK-16873: Channel hangs up when redirected through CLI or AMI |
ASTERISK-16874: [patch] CDR's being written on caller hangup |
ASTERISK-16875: chan_sip returns 503 Service Unavailable response, when additional information would allow it to return 480 response |
ASTERISK-16876: Garbled Audio from IAX trunk |
ASTERISK-16877: [patch] "sip show channel" does not show the complete route set but only the first hop |
ASTERISK-16878: "dahdi show channels" missing the "Extension" number |
ASTERISK-16879: [patch] FollowMe has a maximum of 90 chars for number |
ASTERISK-16880: core show codecs outputs garbage |
ASTERISK-16881: EWS Write Causes Segfault |
ASTERISK-16882: [patch] Missing colon in To/From headers of RTCP manager events |
ASTERISK-16883: Registration failed when remote device sends port number afer domain in From field. |
ASTERISK-16884: [patch] add Path header support to chan_sip |
ASTERISK-16885: [patch] DECT channel driver |
ASTERISK-16886: [patch] Segfault with new meetme function (count callers in menu8) |
ASTERISK-16887: [patch] Events to let know which side (including CLI) of the call hangs up : HangupRequest and SoftHangupRequest |
ASTERISK-16888: CONNECTEDLINE is not working at all if do not introduce a simple SIP call |
ASTERISK-16889: SIP 180 Sent from provider, Asterisk sends 180 and 183 w/SDP. No ringback on calls |
ASTERISK-16890: [patch] Update for chan_unistim fuctionality |
ASTERISK-16891: [regression] Redirect function (over console or AMI) does not work anymore |
ASTERISK-16892: [patch] Asterisk gets killed during the live calling |
ASTERISK-16893: conflicting types for 'locale_t' |
ASTERISK-16894: billsec is always zero when using Local channels |
ASTERISK-16895: Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files: cannot modify limit: Operation not permitted |
ASTERISK-16896: Crash when loading xml documentation |
ASTERISK-16897: [patch] can't have a conversation between two users if the only allowed codec is speex16. |
ASTERISK-16898: SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -> 0 |
ASTERISK-16899: ast_add_hint deadlock in pbx.c MUTEX ast_hint_state_changed |
ASTERISK-16900: [patch] VoicemailMain Exits Without Warning |
ASTERISK-16901: DISA "Cannot handle frames in gsm format" |
ASTERISK-16902: [patch] Random segfault when querying MySQL via func_odbc |
ASTERISK-16903: [patch] adding the CALENDARSTATUS for CALENDAR_WRITE() |
ASTERISK-16904: [patch] adding the CLI: calendar show types |
ASTERISK-16905: Asterisk crashes on an incoming call from an automatic created SIP peer (autocreatepeer=yes) |
ASTERISK-16906: [patch] sippeers command in AMI has incorrect output |
ASTERISK-16907: codecs.conf configuration for speex |
ASTERISK-16908: [regression] Playback does not play the video file if available (.h263) |
ASTERISK-16909: [patch] Realtime field 'fullcontact' populated with invalid data |
ASTERISK-16910: Newstate event contains CallerIDName/CallerIDNum of queue member |
ASTERISK-16911: [regression] INVITEs forwarded to port 5060 instead of real port |
ASTERISK-16912: Attended transfer failure |
ASTERISK-16913: MXML "ajax-response" is not well terminated |
ASTERISK-16914: Intermittent loss of IP address of SIP peer prevents outbound calls |
ASTERISK-16915: Picked call is hanged after 30 seconds (with BYE, because of timeout) |
ASTERISK-16916: TLS working but for SRTP its giving Error |
ASTERISK-16917: Asterisk stop processing any SIP request |
ASTERISK-16918: [patch] CallerID spill to TDM800P FXS port is not sent - possibly NULL |
ASTERISK-16919: [patch] No MOH When Call Parked |
ASTERISK-16920: after update 1.6.2.2 to 1.8.0 registration from peers fails |
ASTERISK-16921: [patch] Dail L(x) shows "Setting call duration limit to x seconds" in CLI |
ASTERISK-16922: Playback/Background stops after 4 or less seconds |
ASTERISK-16923: CDR fields not written after call is bridged |
ASTERISK-16924: AGI playback has bug |
ASTERISK-16925: [patch] distributed device state does not work |
ASTERISK-16926: [patch] Voicemail problems when using only ogg. |
ASTERISK-16927: atxfer fails to read data if Dial exetes macro or gosub |
ASTERISK-16928: [patch] "sip notify clear-mwi" needs terminating CRLF |
ASTERISK-16929: [patch] streamplayer-like utility, but for anything that comes out of a shell pipe |
ASTERISK-16930: [patch] Use ast_sockaddr_stringify_fmt wrappers for sip show peers |
ASTERISK-16931: [patch] ipv6 support in chan_ooh323 |
ASTERISK-16932: [patch] res_odbc using ^ separator in M(macro) option as argument separator in Dial app gives control characters |
ASTERISK-16933: EID is not copied when event is received from XMPP and forwarded into Asterisk's core |
ASTERISK-16934: Asterisk Crash after get many warrring and error on chan_h323.c and RTP |
ASTERISK-16935: [patch] better debug message in devicestate.c |
ASTERISK-16936: [patch] Manager eventfilter: black filters aren't applied properly |
ASTERISK-16937: [patch] devicestate is not aggregated from all received events |
ASTERISK-16938: [patch] Manager Command 'SendText' reports failure on SIP channel |
ASTERISK-16939: Unattended transfer failure |
ASTERISK-16940: Error when storing queue_log data in RT - "column 'time' cannot be type 'int(10) unsigned'" |
ASTERISK-16941: Dropping incompatible voice frame on Local/ on multiple atxfer a->b->c...->d... |
ASTERISK-16942: MutlicastRTP makes asterisk crash |
ASTERISK-16943: [patch] When using Realtime gateway definitions, random crashes occur |
ASTERISK-16944: [patch] Removed resequencing of mailbox |
ASTERISK-16945: AMI command stopMonitor followed by Monitor do not create a new file, just change the name of the original file and append to it |
ASTERISK-16946: [patch] Call to SQLDescribeCol returns an invalid ColumnSize paramenter on x64 (Patch included) |
ASTERISK-16947: Asterisk do not respect priority between queued calls. |
ASTERISK-16948: [patch] MeetMe feature for caching name recordings |
ASTERISK-16949: [patch] Incorrect handling of non-ascii characters in Display Name |
ASTERISK-16950: Asterisk not send fax to fax extension |
ASTERISK-16951: [patch] configure.ac in 1.4.37 broken with autoconf 2.60 |
ASTERISK-16952: [patch] [regression] Change in revision 284478 causes configure to exit when cross-compiling |
ASTERISK-16953: Asterisk 1.6.2.14 w/ Realtime SIP Peers: MWI gets stuck in a loop sending 1000's of SIP NOTIFY messages when rtcachefriends = no |
ASTERISK-16954: Caller Name does not preserve multiple spaces |
ASTERISK-16955: chan_gtalk does not respect the rtp port range configured in rtp.conf |
ASTERISK-16956: asterisk cannot find columns in pgsql realtime database |
ASTERISK-16957: [patch] Asterisk does not play wav files with unknown chunk types |
ASTERISK-16958: Random Crash |
ASTERISK-16959: [regression] sslbindport/tlsbindport in http.conf not working |
ASTERISK-16960: [patch] Core dumped after bridged call is hungup when mysql cdr backend is not available |
ASTERISK-16961: [patch] hint state changes deadlock/race |
ASTERISK-16962: Binary Addon Packages Incompatible |
ASTERISK-16963: Memory leak in asterisk |
ASTERISK-16964: Asterisk does not send release message when channel requested during SETUP gets changed during Procedding Message from TELCO |
ASTERISK-16965: Playback() on a meetme/confbridge stutters/drops randomly |
ASTERISK-16966: [patch] Asterisk segfaults on /lib/i686/cmov/libc.so.6 |
ASTERISK-16967: dropped feature in dial for sending DTMF when progress is made |
ASTERISK-16968: Terrible jitter on SIP->"Fax for Asterisk", SIP->"Playtones", SIP->"MOH" |
ASTERISK-16969: It crashes all the time when I place a call |
ASTERISK-16970: queue.log is inconsistently using interface or membername |
ASTERISK-16971: Playback() on a meetme/confbridge stutters/drops randomly |
ASTERISK-16972: Redirect two bridged channels to the same conference |
ASTERISK-16973: monitor action with "mix: true" does not mix |
ASTERISK-16974: [patch] Redirect with extra channel arg |
ASTERISK-16975: Segmentation Fault: #0 0x080d5dcb in ast_readaudio_callback (s=0xb2e1fa98) at file.c:762 |
ASTERISK-16976: Segfault in sip_get_codec - chan_sip |
ASTERISK-16977: Segfault in handle_response_invite - chan_sip |
ASTERISK-16978: [patch] Translations from SIP 480 responses to cause codes, esp. Do Not Disturb |
ASTERISK-16979: The musiconhold stop when a call is parked |
ASTERISK-16980: ODBC Disconnection if call not briged |
ASTERISK-16981: Voicemail notification - Incorrect VM Duration |
ASTERISK-16982: Fix reconnecting to pgsql database after connection loss. |
ASTERISK-16983: Peer (type=peer) matching is wrong when several peers use one IP:port |
ASTERISK-16984: [patch] T.38 only negotiate 2400 Baud for Faxes |
ASTERISK-16985: [branch] No response is received if we try to subscribe for call completion after we have received a 180 Ringing |
ASTERISK-16986: (Call Completion / SIP) Asterisk Server Fails To Send A Response To A Re-Subscribe After The Expires Timer Matures |
ASTERISK-16987: [patch] (Call Completion / SIP) INVITE Fails (Receive a 404 From Asterisk Server) When Using The URI Provided From A NOTIFY(cc-r |
ASTERISK-16988: (Call Completion / SIP) Displaying Information After We Send A INVITE With URI From A NOTIFY(cc-ready) |
ASTERISK-16989: Transfer = Yes not working |
ASTERISK-16990: Setting CDR(userfield) through features.conf fails |
ASTERISK-16991: [patch] chan_sip continues to call add_peer_mwi_subs which continues to increament MWI NOTIFY's |
ASTERISK-16992: [patch] [regression] meetme conf_run leaks refs |
ASTERISK-16993: regression improper sip parse when invite contains values to the left of the @ ;phone-context=+1;npdi=yes |
ASTERISK-16994: Multi parking blind transfers issue |
ASTERISK-16995: Meetme / auto-call doesn't detect call hangup from voicemail app and keeps meeting open forever |
ASTERISK-16996: ooh323 crashes with segmentation fault every 2-3 days |
ASTERISK-16997: [patch] No answer to OPTIONS packet because Asterisk not looking for 's' in default context |
ASTERISK-16998: AGI CDR Update bug |
ASTERISK-16999: [patch] asterisk crash when setting outbound mwi subscription in sip.conf |